Replace rtc::cricket::Settable with rtc::Maybe

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 470 matching lines @@
   // No automatic resizing when using simulcast or screencast.
   bool automatic_resize =
       !is_screencast && parameters_.config.rtp.ssrcs.size() == 1;
   bool frame_dropping = !is_screencast;
   bool denoising;
   bool codec_default_denoising = false;
   if (is_screencast) {
     denoising = false;
   } else {
     // Use codec default if video_noise_reduction is unset.
-    codec_default_denoising = !options.video_noise_reduction.Get(&denoising);
+    codec_default_denoising = !options.video_noise_reduction;
+    denoising = options.video_noise_reduction.value_or(false);
   }
 
   if (CodecNamesEq(codec.name, kVp8CodecName)) {
     encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
     encoder_settings_.vp8.automaticResizeOn = automatic_resize;
     // VP8 denoising is enabled by default.
     encoder_settings_.vp8.denoisingOn =
         codec_default_denoising ? true : denoising;
     encoder_settings_.vp8.frameDroppingOn = frame_dropping;
     return &encoder_settings_.vp8;
@@ skipping 267 matching lines @@
     const std::vector<VideoCodec>& recv_codecs,
     WebRtcVideoEncoderFactory* external_encoder_factory,
     WebRtcVideoDecoderFactory* external_decoder_factory)
     : call_(call),
       unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
       external_encoder_factory_(external_encoder_factory),
       external_decoder_factory_(external_decoder_factory) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   SetDefaultOptions();
   options_.SetAll(options);
-  options_.cpu_overuse_detection.Get(&signal_cpu_adaptation_);
+  if (options_.cpu_overuse_detection)
+    signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
   rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
   sending_ = false;
   default_send_ssrc_ = 0;
   SetRecvCodecs(recv_codecs);
 }
 
 void WebRtcVideoChannel2::SetDefaultOptions() {
-  options_.cpu_overuse_detection.Set(true);
-  options_.dscp.Set(false);
-  options_.suspend_below_min_bitrate.Set(false);
-  options_.screencast_min_bitrate.Set(0);
+  options_.cpu_overuse_detection = true;
+  options_.dscp = false;
+  options_.suspend_below_min_bitrate = false;
+  options_.screencast_min_bitrate = 0;
 }
 
 WebRtcVideoChannel2::~WebRtcVideoChannel2() {
   for (auto& kv : send_streams_)
     delete kv.second;
   for (auto& kv : receive_streams_)
     delete kv.second;
 }
 
 bool WebRtcVideoChannel2::CodecIsExternallySupported(
@@ skipping 143 matching lines @@
   const std::vector<VideoCodecSettings> supported_codecs =
       FilterSupportedCodecs(MapCodecs(codecs));
 
   if (supported_codecs.empty()) {
     LOG(LS_ERROR) << "No video codecs supported.";
     return false;
   }
 
   LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString();
 
-  VideoCodecSettings old_codec;
-  if (send_codec_.Get(&old_codec) && supported_codecs.front() == old_codec) {
+  if (send_codec_ && supported_codecs.front() == *send_codec_) {
     LOG(LS_INFO) << "Ignore call to SetSendCodecs because first supported "
                     "codec hasn't changed.";
     // Using same codec, avoid reconfiguring.
     return true;
   }
 
-  send_codec_.Set(supported_codecs.front());
+  send_codec_ = supported_codecs.front();
 
   rtc::CritScope stream_lock(&stream_crit_);
   LOG(LS_INFO) << "Change the send codec because SetSendCodecs has a different "
                   "first supported codec.";
   for (auto& kv : send_streams_) {
     RTC_DCHECK(kv.second != nullptr);
     kv.second->SetCodec(supported_codecs.front());
   }
   LOG(LS_INFO) << "SetNackAndRemb on all the receive streams because the send "
                   "codec has changed.";
@@ skipping 25 matching lines @@
     bitrate_config_.max_bitrate_bps = bitrate_kbps * 1000;
   } else {
     bitrate_config_.max_bitrate_bps = -1;
   }
   call_->SetBitrateConfig(bitrate_config_);
 
   return true;
 }
 
 bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
-  VideoCodecSettings codec_settings;
-  if (!send_codec_.Get(&codec_settings)) {
+  if (!send_codec_) {
     LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
     return false;
   }
-  *codec = codec_settings.codec;
+  *codec = send_codec_->codec;
   return true;
 }
 
 bool WebRtcVideoChannel2::SetSendStreamFormat(uint32_t ssrc,
                                               const VideoFormat& format) {
   LOG(LS_VERBOSE) << "SetSendStreamFormat:" << ssrc << " -> "
                   << format.ToString();
   rtc::CritScope stream_lock(&stream_crit_);
   if (send_streams_.find(ssrc) == send_streams_.end()) {
     return false;
   }
   return send_streams_[ssrc]->SetVideoFormat(format);
 }
 
 bool WebRtcVideoChannel2::SetSend(bool send) {
   LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
-  if (send && !send_codec_.IsSet()) {
+  if (send && !send_codec_) {
     LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
     return false;
   }
   if (send) {
     StartAllSendStreams();
   } else {
     StopAllSendStreams();
   }
   sending_ = send;
   return true;
@@ skipping 175 matching lines @@
 
   for (uint32_t used_ssrc : sp.ssrcs)
     receive_ssrcs_.insert(used_ssrc);
 
   webrtc::VideoReceiveStream::Config config(this);
   ConfigureReceiverRtp(&config, sp);
 
   // Set up A/V sync group based on sync label.
   config.sync_group = sp.sync_label;
 
-  config.rtp.remb = false;
-  VideoCodecSettings send_codec;
-  if (send_codec_.Get(&send_codec)) {
-    config.rtp.remb = HasRemb(send_codec.codec);
-  }
+  config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
 
   receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
       call_, sp, config, external_decoder_factory_, default_stream,
       recv_codecs_);
 
   return true;
 }
 
 void WebRtcVideoChannel2::ConfigureReceiverRtp(
     webrtc::VideoReceiveStream::Config* config,
@@ skipping 362 matching lines @@
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
   LOG(LS_INFO) << "SetOptions: " << options.ToString();
   VideoOptions old_options = options_;
   options_.SetAll(options);
   if (options_ == old_options) {
     // No new options to set.
     return true;
   }
   {
     rtc::CritScope lock(&capturer_crit_);
-    options_.cpu_overuse_detection.Get(&signal_cpu_adaptation_);
+    if (options_.cpu_overuse_detection)
+      signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
   }
-  rtc::DiffServCodePoint dscp = options_.dscp.GetWithDefaultIfUnset(false)
-                                    ? rtc::DSCP_AF41
-                                    : rtc::DSCP_DEFAULT;
+  rtc::DiffServCodePoint dscp =
+      options_.dscp.value_or(false) ? rtc::DSCP_AF41 : rtc::DSCP_DEFAULT;
   MediaChannel::SetDscp(dscp);
   rtc::CritScope stream_lock(&stream_crit_);
   for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
            send_streams_.begin();
        it != send_streams_.end(); ++it) {
     it->second->SetOptions(options_);
   }
   return true;
 }
 
@@ skipping 71 matching lines @@
        it != send_streams_.end(); ++it) {
     it->second->Stop();
   }
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
     VideoSendStreamParameters(
         const webrtc::VideoSendStream::Config& config,
         const VideoOptions& options,
         int max_bitrate_bps,
-        const Settable<VideoCodecSettings>& codec_settings)
+        const rtc::Maybe<VideoCodecSettings>& codec_settings)
     : config(config),
       options(options),
       max_bitrate_bps(max_bitrate_bps),
-      codec_settings(codec_settings) {
-}
+      codec_settings(codec_settings) {}
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder(
     webrtc::VideoEncoder* encoder,
     webrtc::VideoCodecType type,
     bool external)
     : encoder(encoder),
       external_encoder(nullptr),
       type(type),
       external(external) {
   if (external) {
     external_encoder = encoder;
     this->encoder =
         new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
   }
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
     webrtc::Call* call,
     const StreamParams& sp,
     const webrtc::VideoSendStream::Config& config,
     WebRtcVideoEncoderFactory* external_encoder_factory,
     const VideoOptions& options,
     int max_bitrate_bps,
-    const Settable<VideoCodecSettings>& codec_settings,
+    const rtc::Maybe<VideoCodecSettings>& codec_settings,
     const std::vector<webrtc::RtpExtension>& rtp_extensions)
     : ssrcs_(sp.ssrcs),
       ssrc_groups_(sp.ssrc_groups),
       call_(call),
       external_encoder_factory_(external_encoder_factory),
       stream_(NULL),
       parameters_(config, options, max_bitrate_bps, codec_settings),
       allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
       capturer_(NULL),
       sending_(false),
       muted_(false),
       old_adapt_changes_(0),
       first_frame_timestamp_ms_(0),
       last_frame_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
   parameters_.config.rtp.c_name = sp.cname;
   parameters_.config.rtp.extensions = rtp_extensions;
 
-  VideoCodecSettings params;
-  if (codec_settings.Get(&params)) {
-    SetCodec(params);
+  if (codec_settings) {
+    SetCodec(*codec_settings);
   }
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
   DisconnectCapturer();
   if (stream_ != NULL) {
     call_->DestroyVideoSendStream(stream_);
   }
   DestroyVideoEncoder(&allocated_encoder_);
 }
@@ skipping 157 matching lines @@
   rtc::CritScope cs(&lock_);
   if (capturer_ == NULL)
     return;
 
   capturer_->SetApplyRotation(apply_rotation);
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
     const VideoOptions& options) {
   rtc::CritScope cs(&lock_);
-  VideoCodecSettings codec_settings;
-  if (parameters_.codec_settings.Get(&codec_settings)) {
+  if (parameters_.codec_settings) {
     LOG(LS_INFO) << "SetCodecAndOptions because of SetOptions; options="
                  << options.ToString();
-    SetCodecAndOptions(codec_settings, options);
+    SetCodecAndOptions(*parameters_.codec_settings, options);
   } else {
     parameters_.options = options;
   }
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
     const VideoCodecSettings& codec_settings) {
   rtc::CritScope cs(&lock_);
   LOG(LS_INFO) << "SetCodecAndOptions because of SetCodec.";
   SetCodecAndOptions(codec_settings, parameters_.options);
@@ skipping 84 matching lines @@
                          "payload type. Ignoring.";
       parameters_.config.rtp.rtx.ssrcs.clear();
     } else {
       parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
     }
   }
 
   parameters_.config.rtp.nack.rtp_history_ms =
       HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
 
-  options.suspend_below_min_bitrate.Get(
-      &parameters_.config.suspend_below_min_bitrate);
+  RTC_CHECK(options.suspend_below_min_bitrate);
+  parameters_.config.suspend_below_min_bitrate =
+      *options.suspend_below_min_bitrate;
 
-  parameters_.codec_settings.Set(codec_settings);
+  parameters_.codec_settings = codec_settings;
   parameters_.options = options;
 
   LOG(LS_INFO)
       << "RecreateWebRtcStream (send) because of SetCodecAndOptions; options="
       << options.ToString();
   RecreateWebRtcStream();
   if (allocated_encoder_.encoder != new_encoder.encoder) {
     DestroyVideoEncoder(&allocated_encoder_);
     allocated_encoder_ = new_encoder;
   }
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions(
     const std::vector<webrtc::RtpExtension>& rtp_extensions) {
   rtc::CritScope cs(&lock_);
   parameters_.config.rtp.extensions = rtp_extensions;
   if (stream_ != nullptr) {
     LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetRtpExtensions";
     RecreateWebRtcStream();
   }
 }
 
 webrtc::VideoEncoderConfig
 WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
     const Dimensions& dimensions,
     const VideoCodec& codec) const {
   webrtc::VideoEncoderConfig encoder_config;
   if (dimensions.is_screencast) {
-    int screencast_min_bitrate_kbps;
-    parameters_.options.screencast_min_bitrate.Get(
-        &screencast_min_bitrate_kbps);
+    RTC_CHECK(parameters_.options.screencast_min_bitrate);
     encoder_config.min_transmit_bitrate_bps =
-        screencast_min_bitrate_kbps * 1000;
+        *parameters_.options.screencast_min_bitrate * 1000;
     encoder_config.content_type =
         webrtc::VideoEncoderConfig::ContentType::kScreen;
   } else {
     encoder_config.min_transmit_bitrate_bps = 0;
     encoder_config.content_type =
         webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
   }
 
   // Restrict dimensions according to codec max.
   int width = dimensions.width;
@@ skipping 15 matching lines @@
   size_t stream_count = parameters_.config.rtp.ssrcs.size();
   if (IsCodecBlacklistedForSimulcast(codec.name) || dimensions.is_screencast) {
     stream_count = 1;
   }
 
   encoder_config.streams =
       CreateVideoStreams(clamped_codec, parameters_.options,
                          parameters_.max_bitrate_bps, stream_count);
 
   // Conference mode screencast uses 2 temporal layers split at 100kbit.
-  if (parameters_.options.conference_mode.GetWithDefaultIfUnset(false) &&
+  if (parameters_.options.conference_mode.value_or(false) &&
       dimensions.is_screencast && encoder_config.streams.size() == 1) {
     ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
 
     // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
     // on the VideoCodec struct as target and max bitrates, respectively.
     // See eg. webrtc::VP8EncoderImpl::SetRates().
     encoder_config.streams[0].target_bitrate_bps =
         config.tl0_bitrate_kbps * 1000;
     encoder_config.streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
     encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
@@ skipping 14 matching lines @@
   }
   LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
                << (is_screencast ? " (screencast)" : " (not screencast)");
 
   last_dimensions_.width = width;
   last_dimensions_.height = height;
   last_dimensions_.is_screencast = is_screencast;
 
   RTC_DCHECK(!parameters_.encoder_config.streams.empty());
 
-  VideoCodecSettings codec_settings;
-  parameters_.codec_settings.Get(&codec_settings);
+  RTC_CHECK(parameters_.codec_settings);
+  VideoCodecSettings codec_settings = *parameters_.codec_settings;
 
   webrtc::VideoEncoderConfig encoder_config =
       CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
 
   encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
       codec_settings.codec, parameters_.options, is_screencast);
 
   bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
 
   encoder_config.encoder_specific_settings = NULL;
@@ skipping 24 matching lines @@
 
 VideoSenderInfo
 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
   VideoSenderInfo info;
   webrtc::VideoSendStream::Stats stats;
   {
     rtc::CritScope cs(&lock_);
     for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
       info.add_ssrc(ssrc);
 
-    VideoCodecSettings codec_settings;
-    if (parameters_.codec_settings.Get(&codec_settings))
-      info.codec_name = codec_settings.codec.name;
+    if (parameters_.codec_settings)
+      info.codec_name = parameters_.codec_settings->codec.name;
     for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
       if (i == parameters_.encoder_config.streams.size() - 1) {
         info.preferred_bitrate +=
             parameters_.encoder_config.streams[i].max_bitrate_bps;
       } else {
         info.preferred_bitrate +=
             parameters_.encoder_config.streams[i].target_bitrate_bps;
       }
     }
 
@@ skipping 91 matching lines @@
   int width = last_dimensions_.width;
   last_dimensions_.width = 0;
   SetDimensions(width, last_dimensions_.height, last_dimensions_.is_screencast);
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
   if (stream_ != NULL) {
     call_->DestroyVideoSendStream(stream_);
   }
 
-  VideoCodecSettings codec_settings;
-  parameters_.codec_settings.Get(&codec_settings);
+  RTC_CHECK(parameters_.codec_settings);
   parameters_.encoder_config.encoder_specific_settings =
       ConfigureVideoEncoderSettings(
-          codec_settings.codec, parameters_.options,
+          parameters_.codec_settings->codec, parameters_.options,
           parameters_.encoder_config.content_type ==
               webrtc::VideoEncoderConfig::ContentType::kScreen);
 
   webrtc::VideoSendStream::Config config = parameters_.config;
   if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {
     LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
                        "payload type the set codec. Ignoring RTX.";
     config.rtp.rtx.ssrcs.clear();
   }
   stream_ = call_->CreateVideoSendStream(config, parameters_.encoder_config);
@@ skipping 417 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO